search for: channelid

...Jun 17, 2023 at 2:55 PM TTT <lists at telium.io> wrote: > Based on postings it should be possible to get the SIP Call-ID header > value from the ARI. At what point is this value available ? As well, how > do I retrieve that value – something like > > > > GET /channels/{channelId}/pjsip_header?key=Call-Id > > > > But that doesn’t work. > 'pjsip_header' is not a valid route. All possible routes are documented on the wiki, if it's not there then it doesn't exist. Instead you would use variable[1] to execute the PJSIP_HEADER dialplan function[2...

I tried GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id) But it responds with "message": "Channel not in Stasis application" Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan,...

Based on postings it should be possible to get the SIP Call-ID header value from the ARI. At what point is this value available ? As well, how do I retrieve that value - something like GET /channels/{channelId}/pjsip_header?key=Call-Id But that doesn't work. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230617/ae7c536c/attachment.html>

Hi all, i have observed, that sometimes all BChannels on my Zaptel Pri device (E400P) will be restarted. The E400P is connected to another pri switch. In the traces from the other side (pri switch) i can see that libpri request for the channelid is 255. Is this a bug or a feature ...? Or, can it be a bug on the other side (terminator switch) ? Have anyone an idea ? Thanks, Thomas.

...erisk Channel Source include the Speex Library in resample this frame in 32KHz to 8KHz. Searching for it in Speex Doc, I found it: SpeexResamplerState *resampler; resampler = speex_resampler_init(nb_channels, input_rate, output_rate, quality, &err); err = speex_resampler_process_int(resampler, channelID, in, &in_length, out, &out_length); But in my source I have one *dataframe where is my payload. How I can use it to resample my frame? Thanks you very much, Thiago. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/at...

Hi all, I am using Speex in a VoIP application and everything is working great. Now I am trying to integrate the Resampler in order to convert data input, especially in the Wide Band mode (16 Khz). I have seen in the doc that for mono the ChannelID parameter should be 0, but How one should call the resampler in the case of PCM stereo data ? It is also stated that "It is also possible to process multiple channels at once" but I did not find further documentation (I am using the Fixed Point implementation). So How to do it ? Thanks...

...of a single audio packet/frame > through the system. Let's try to sum it up on base of the given easy example how to get the complete delay between those two speakers: A calls B: ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT.... =========================================================================================================== c8137221 327-00000004 03:22:42 g722 608K 0 0 0.000 608K 0 0 0.0...

..........................> <Status....> <RTT(ms)..> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <MatchList.................................................................> Channel: <ChannelId......................................> <State.....> <Time(sec)> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> ========================================================================================= Endpoint: 9001...

.....> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.........................................................> Match: <ip/cidr.........................> Channel: <ChannelId......................................> <State.....> <Time(sec)> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> ========================================================================================= Endpoint: demo-alice Unavailable 0 o...

.....> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.......................................... ...............> Match: <ip/cidr.........................> Channel: <ChannelId......................................> <State.....> <Time(sec)> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> =========================================================== ============================== Endpoint: murftest12/101...

Hello! I'm just wondering what the RTT exactly means. Where are the exact measuring points located? > pjsip show channelstats ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT.... =========================================================================================================== c8137221 327-00000004 03:22:42 g722 608K 0 0 0.000 608K 0 0 0.0...

...annel Source include the Speex Library in resample this frame in 32KHz to 8KHz. Searching for it in Speex Doc, I found it: SpeexResamplerState *resampler; resampler = speex_resampler_init(nb_channels, input_rate, output_rate, quality, &err); err = speex_resampler_process_int(resampler, channelID, in, &in_length, out, &out_length); But in my source I have one *dataframe where is my payload. How I can use it to resample my frame? Thanks you very much, Thiago. ------------------------------------------------------------------------------ __________________________________...

...socket leak on REGISTER SIP messages. We've seen it happen only on customers using Sipura SPA2100 ATAs. If I issue a "sip show channels", I see thousands of "zombie channels". If I look into the details, that's what I get - actually one single "sip show channel <channelID>" returns thousands of these: * SIP Call Direction: Incoming Call-ID: 5c2cf755-ccde1b6c@x.x.x.xq: 520 REGISTER Our Codec Capability: 12 Non-Codec Capability: 1 Their Codec Capability: 0 Joint Codec Capability: 0 Format un...

...think ${BRIDGEPEER} will work only when both channels are connected. i want to get channel-id before dialing so that i can dial using that channel id. > ${BRIDGEPEER} is probably a good way to do what you want.. if Channel > A calls Channel B, and you want Channel A to "get" the channelID of > Channel B, as long as the two channels are bridged, ${BRIDGEPEER} will > do what you want >> perhaps ${CALLERID(DNID)} >> >>> my question is how can i get channel-id of a user or peer. I tried using >>> ChanIsAvail(username). this works correctly when user...

HiCould someone please help me how to set Custom generated UUID in Originate action in AMI ? RegardsBabak -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150508/528d5ff1/attachment.html>

...llowing issues are resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-28375 - res_pjsip: New configuration setting to allow disabling norefersub (Reported by Dan Cropp) * ASTERISK-28320 - Added ARI resource /ari/channels/{channelid}/rtp_statistics (Reported by sungtae kim) Bugs fixed in this release: ----------------------------------- * ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure (Reported by Guido Falsi) * ASTERISK-28412 - GCC 9 catches more...

...gt; <Type> <cos> <tos> >> <BindAddress..................> >> Identify: >> <Identify/Endpoint.........................................................> >> Match: <ip/cidr.........................> >> Channel: <ChannelId......................................> >> <State.....> <Time(sec)> >> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> >> >> ========================================================================================= >&...

...TransportId........> <Type> <cos> <tos> > <BindAddress..................> > Identify: > <Identify/Endpoint.........................................................> > Match: <ip/cidr.........................> > Channel: <ChannelId......................................> > <State.....> <Time(sec)> > Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> > > ========================================================================================= > > Endpoint:...

...llowing issues are resolved in this release: New Features made in this release: ----------------------------------- * ASTERISK-28375 - res_pjsip: New configuration setting to allow disabling norefersub (Reported by Dan Cropp) * ASTERISK-28320 - Added ARI resource /ari/channels/{channelid}/rtp_statistics (Reported by sungtae kim) Bugs fixed in this release: ----------------------------------- * ASTERISK-28427 - new mwi.h include missing from some dahdi source files, causes build failure (Reported by Guido Falsi) * ASTERISK-28412 - GCC 9 catches more...

On 15.05.20 at 14:31 Doug Lytle wrote: > Google says Round Trip Time > > https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points located? => How are the RTT values exactly calculated? Which values are actually